search for: plancommunications

...BRI circuits. You may also want to switch to 100% VoIP provided termination, porting your number to a carrier (I'm recently a fan of NuFone, and they're relatively local to you, with centers in Michigan and Chicago- but YMMV), as you know that's an echo free solution. -Paul Davidson PlanCommunications, LLC Date: Tue, 28 Feb 2006 09:52:02 -0600 From: "Brent Torrenga" <lists@torrenga.com> Subject: [Asterisk-Users] Re: Echo and other reasons to migrate to BRI from POTS? Was (Echo on PRI/BRI?) To: <asterisk-users@lists.digium.com> Message-ID: <000b01c63...

...only, in my opinion, >unless you have other compelling reasons to do so- there's a lot of other >places to look first. >If you could describe more of your particular setup, I'd be happy to give >more detailed description of where the problems might lie. >-Paul Davidson > PlanCommunications, LLC Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of "office-style" phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk

...known to do some non-standard things, and various channel drivers can take offence with that. If SIP is available (CCM 4.0and above), you may want to consider re-architecting to it, as a good many of these problems go away under SIP. Let me know if I can be of further assistance. -Paul Davidson PlanCommunications, LLC. On 1/12/06, asterisk-users-request@lists.digium.com < asterisk-users-request@lists.digium.com> wrote: > > > Message: 4 > Date: Thu, 12 Jan 2006 11:01:02 -0500 > From: "Peckham, Christopher" <CPeckham@mercy.edu> > Subject: [Asterisk-Users] Transfer issu...

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from

Hey Everyone, We are in the works of planning a new * installation for our company. We have 20 users in our main office and 5 users in a remote office a couple of states away. Our call volume for the main office will be anywhere from 5-10 concurrent calls. The remote office will have about 3 heavy users with two users making calls occasionally. Right now we have an existing PBX. We have a